The present invention relates generally to voice communication systems and, more particularly, to systems for transmitting and receiving voice information over packet-switched networks.
For years, the telecommunications industry has examined was to combine the flexibility and functionality of packet-switched networks primarily used for transmitting data (e.g., the Internet) with the accuracy and speed of conventional circuit based telephone networks (i.e., the Public Switched Telephone Network or PSTN). Conventional telephone systems differ from modern data-based computer networks in several ways. Most importantly however, are the differences in how connections between the sender and the recipient are made.
In conventional telephone systems, when a caller picks up his telephone, an OFFHOOK message is sent from the phone across the PSTN to the user's central office (CO). In response, the CO sends a dialtone back to the user's phone indicated that he is connected and can initiate a call. Next, the caller dials the phone number of the intended recipient and, through the keypad tones or pulses, this information is transmitted to the CO. In response, the CO transmits a RINGING message causing the recipient's phone to ring. If the recipient picks up the phone, the recipient's phone sends an OFFHOOK message to the CO and a dedicated circuit across the PSTN between the caller and the recipient is established, enabling voice traffic to pass between the connected parties in a smooth, seamless manner. Typically, the voice traffic is digitized at the CO and transmitted over the dedicated PSTN circuit using a technology called time division multiplexing (TDM). This dedicated circuit continuously transmits information between the parties at a rate of about 128 kilobits per second (kbps) (64 kbps each way) for the duration of the call. For a five minute telephone call, this equates to the transmission of approximately 4.7 megabytes (MB) of information.
Unfortunately, in most telephone conversations, much of the bandwidth required to enable the transmission of information between the parties is wasted. For example, because people typically do not speak while the other party is speaking, almost half of the available bandwidth is wasted during the call. Similarly, during periods of silence (even milliseconds at a time), no information needs to pass between the parties. However, because of the dedicated, physical circuit between the parties, information is passed regardless of content.
Contrary to conventional telephone systems, most data networks such as the Internet, do not transmit information across dedicated, physical circuits. Rather, information sent between two computers on a network is broken up in a series of small packets. These packets are then routed to the destination and reassembled at the recipient end. Various protocols have been developed for enabling the efficient and accurate transfer of information across computer networks, such as interne protocol (IP), asynchronous transfer mode (ATM), Ethernet, etc. Because computer networks only transmit the information which needs to be relayed, there is little wasted bandwidth.
Because of the rising need for network bandwidth and the continued need to optimize bandwidth which is already available, efforts have been made to reduce the bandwidth cost of voice traffic by routing voice traffic over packet-switched networks. This concept is generally referred to as voice over packet telephony (, e.g., VoIP), although various other transmission methods and network protocols may also be employed, such as DSL, ATM, or the like. In general, the concept of VoIP requires a seamless experience on the part of the user. That is, conventional telephone systems (referred to as plain old telephone systems or POTS) must be able to utilize the technology in an invisible manner. In practice, similar to conventional PSTN devices, when a POTS device (or analogous customer premises equipment (CPE) device) goes off hook, a message is sent to a CO indicating this state. A dialed number is then received by the CO, indicating the recipient's address, and the corresponding voice traffic is digitized and packetized at the CO for transmission to the recipient's CPE device.
To assist in enabling the effective use of VoIP technology, many CPE devices include support for simultaneous operation of both VoIP and POTS systems. In such a system, a splitter device located at the CO operates to separate received and transmitted signals from the CPE. Upon receipt of an incoming POTS call, signals are first passed through a surge suppressor device which operates to clip the ringing signals. Unfortunately, during ringing on the POTS line, the distortion generated by the surge suppressor is often larger than a transmitted data signal, resulting in a loss of data transmission if a VoIP call is underway during the ringing. Following this disruption in service, it is often necessary to retrain the CPE VoIP system. This process is time consuming (i.e., ˜11 secs.) and results in disruption of the outgoing call.
Accordingly, it is desired to provide a method and system for preventing the disruption of VoIP calls caused by concurrent ringing on a POTS line.